• Title/Summary/Keyword: Signal representation

Search Result 181, Processing Time 0.032 seconds

Performance Comparison of BCS-SPL Techniques Against a Variety of Restoring Block Sizes (복원 블록 크기 변화에 따른 BCS-SPL기법의 이미지 복원 성능 비교)

  • Ryu, Joong-seon;Kim, Jin-soo
    • Journal of Korea Society of Industrial Information Systems
    • /
    • v.21 no.3
    • /
    • pp.21-28
    • /
    • 2016
  • Compressed sensing is a signal processing technique for efficiently acquiring and reconstructing in an under-sampled (i.e., under Nyquist rate) representation. Specially, a block compressed sensing with Smoothed Projected Landweber (BCS-SPL) framework is one of the most widely used schemes. Currently, a variety of BCS-SPL schemes have been actively studied. However, when restoring, block sizes have effects on the reconstructed visual qualities, and in this paper, both a basic scheme of BCS-SPL and several modified schemes of BCS-SPL with structured measurement matrix are analyzed for the effects of the block sizes on the performances of reconstructed image qualities. Through several experiments, it is shown that a basic scheme of BCS-SPL provides superior performance in block size 4.

AM-FM Decomposition and Estimation of Instantaneous Frequency and Instantaneous Amplitude of Speech Signals for Natural Human-robot Interaction (자연스런 인간-로봇 상호작용을 위한 음성 신호의 AM-FM 성분 분해 및 순간 주파수와 순간 진폭의 추정에 관한 연구)

  • Lee, He-Young
    • Speech Sciences
    • /
    • v.12 no.4
    • /
    • pp.53-70
    • /
    • 2005
  • A Vowel of speech signals are multicomponent signals composed of AM-FM components whose instantaneous frequency and instantaneous amplitude are time-varying. The changes of emotion states cause the variation of the instantaneous frequencies and the instantaneous amplitudes of AM-FM components. Therefore, it is important to estimate exactly the instantaneous frequencies and the instantaneous amplitudes of AM-FM components for the extraction of key information representing emotion states and changes in speech signals. In tills paper, firstly a method decomposing speech signals into AM - FM components is addressed. Secondly, the fundamental frequency of vowel sound is estimated by the simple method based on the spectrogram. The estimate of the fundamental frequency is used for decomposing speech signals into AM-FM components. Thirdly, an estimation method is suggested for separation of the instantaneous frequencies and the instantaneous amplitudes of the decomposed AM - FM components, based on Hilbert transform and the demodulation property of the extended Fourier transform. The estimates of the instantaneous frequencies and the instantaneous amplitudes can be used for modification of the spectral distribution and smooth connection of two words in the speech synthesis systems based on a corpus.

  • PDF

Monitoring QZSS CLAS-based VRS-RTK Positioning Performance

  • Lim, Cheolsoon;Lee, Yebin;Cha, Yunho;Park, Byungwoon;Park, Sul Gee;Park, Sang Hyun
    • Journal of Positioning, Navigation, and Timing
    • /
    • v.11 no.4
    • /
    • pp.251-261
    • /
    • 2022
  • The Centimeter Level Augmentation Service (CLAS) is the Precise Point Positioning (PPP) - Real Time Kinematic (RTK) correction service utilizing the Quasi-Zenith Satellite System (QZSS) L6 (1278.65 MHz) signal to broadcast the Global Navigation Satellite System (GNSS) error corrections. Compact State-Space Representation (CSSR) corrections for mitigating GNSS measurement error sources such as satellite orbit, clock, code and phase biases, tropospheric error, ionospheric error are estimated from the ground segment of QZSS CLAS using the code and carrier-phase measurements collected in the Japan's GNSS Earth Observation Network (GEONET). Since the CLAS service begun on November 1, 2018, users with dedicated receivers can perform cm-level precise positioning using CSSR corrections. In this paper, CLAS-based VRS-RTK performance evaluation was performed using Global Positioning System (GPS) observables collected from the refence station, TSK2, located in Japan. As a result of performing GPS-only RTK positioning using the open-source software CLASLIB and RTKLIB, it took about 15 minutes to resolve the carrier-phase ambiguities, and the RTK fix rate was only about 41%. Also, the Root Mean Squares (RMS) values of position errors (fixed only) are about 4cm horizontally and 7 cm vertically.

Evaluation of Single-Frequency Precise Point Positioning Performance Based on SPARTN Corrections Provided by the SAPCORDA SAPA Service

  • Kim, Yeong-Guk;Kim, Hye-In;Lee, Hae-Chang;Kim, Miso;Park, Kwan-Dong
    • Journal of Positioning, Navigation, and Timing
    • /
    • v.10 no.2
    • /
    • pp.75-82
    • /
    • 2021
  • Fields of high-precision positioning applications are growing fast across the mass market worldwide. Accordingly, the industry is focusing on developing methods of applying State-Space Representation (SSR) corrections on low-cost GNSS receivers. Among SSR correction types, this paper analyzes Safe Position Augmentation for Real Time Navigation (SPARTN) messages being offered by the SAfe and Precise CORrection DAta (SAPCORDA) company and validates positioning algorithms based on them. The first part of this paper introduces the SPARTN format in detail. Then, procedures on how to apply Basic-Precision Atmosphere Correction (BPAC) and High-Precision Atmosphere Correction (HPAC) messages are described. BPAC and HPAC messages are used for correcting satellite clock errors, satellite orbit errors, satellite signal biases and also ionospheric and tropospheric delays. Accuracies of positioning algorithms utilizing SPARTN messages were validated with two types of positioning strategies: Code-PPP using GPS pseudorange measurements and PPP-RTK including carrier phase measurements. In these performance checkups, only single-frequency measurements have been used and integer ambiguities were estimated as float numbers instead of fixed integers. The result shows that, with BPAC and HPAC corrections, the horizontal accuracy is 46% and 63% higher, respectively, compared to that obtained without application of SPARTN corrections. Also, the average horizontal and vertical RMSE values with HPAC are 17 cm and 27 cm, respectively.

Revisiting Deep Learning Model for Image Quality Assessment: Is Strided Convolution Better than Pooling? (영상 화질 평가 딥러닝 모델 재검토: 스트라이드 컨볼루션이 풀링보다 좋은가?)

  • Uddin, AFM Shahab;Chung, TaeChoong;Bae, Sung-Ho
    • Proceedings of the Korean Society of Broadcast Engineers Conference
    • /
    • 2020.11a
    • /
    • pp.29-32
    • /
    • 2020
  • Due to the lack of improper image acquisition process, noise induction is an inevitable step. As a result, objective image quality assessment (IQA) plays an important role in estimating the visual quality of noisy image. Plenty of IQA methods have been proposed including traditional signal processing based methods as well as current deep learning based methods where the later one shows promising performance due to their complex representation ability. The deep learning based methods consists of several convolution layers and down sampling layers for feature extraction and fully connected layers for regression. Usually, the down sampling is performed by using max-pooling layer after each convolutional block. We reveal that this max-pooling causes information loss despite of knowing their importance. Consequently, we propose a better IQA method that replaces the max-pooling layers with strided convolutions to down sample the feature space and since the strided convolution layers have learnable parameters, they preserve optimal features and discard redundant information, thereby improve the prediction accuracy. The experimental results verify the effectiveness of the proposed method.

  • PDF

Inverse Estimation of Geoacoustic Parameters in Shallow Water Using tight Bulb Sound Source (천해환경에서 전구음원을 이용한 지음향인자의 역추정)

  • 한주영;이성욱;나정열;김성일
    • The Journal of the Acoustical Society of Korea
    • /
    • v.23 no.1
    • /
    • pp.8-16
    • /
    • 2004
  • An inversion method is presented for the determination of the compressional wave speed, compressional wave attenuation, thickness of the sediment layer and density as a function of depth for a horizontally stratified ocean bottom. An experiment for estimating those properties was conducted in the shallow water of South Sea in Korea. In the experiment, a light bulb implosion and the propagating sound were measured using a VLA (vertical line array). As a method for estimating the geoacoustic properties, a coherent broadband matched field processing combined with Genetic Algorithm was employed. When a time-dependent signal is very short, the Fourier transform results are not accurate, since the frequency components are not locatable in time and the windowed Fourier transform is limited by the length of the window. However, it is possible to do this using the wavelet transform a transform that yields a time-frequency representation of a signal. In this study, this transform is used to identify and extract the acoustic components from multipath time series. The inversion is formulated as an optimization problem which maximizes the cost function defined as a normalized correlation between the measured and modeled signals in the wavelet transform coefficient vector. The experiments and procedures for deploying the light bulbs and the coherent broadband inversion method are described, and the estimated geoacoustic profile in the vicinity of the VLA site is presented.

A Study on ACFBD-MPC in 8kbps (8kbps에 있어서 ACFBD-MPC에 관한 연구)

  • Lee, See-Woo
    • Journal of the Korea Academia-Industrial cooperation Society
    • /
    • v.17 no.7
    • /
    • pp.49-53
    • /
    • 2016
  • Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

A Study on Development of Off-Line Path Programming for Footwear Buffing Robot

  • Lho, Tae-Jung;Kang, Dong-Joon;Che, Woo-Seung;Kim, Jung-Young;Kim, Min-Sung
    • 제어로봇시스템학회:학술대회논문집
    • /
    • 2004.08a
    • /
    • pp.1469-1473
    • /
    • 2004
  • We suggest how to program off-line robot path along shoes' outsole shape in the footwear buffing process by a 5-axis microscribe system like robot arms. This microscribe system developed consists a 5-axis robot link with a turn table, signal processing circuit, PC and an application software program. It makes a robot path on the shoe's upper through the movement of a microscribe with many joints. To do this, first it reads 5-encoder's pulse values while a robot arm points a shoes' outsole shape from the initial status. This system developed calculates the encoder pulse values for the robot arm's rotation and transmits the angle pulse values to the PC through a circuit. Then, Denavit-Hartenberg's(D-H) direct kinematics is used to make the global coordinate from robot joint one. The determinant is obtained with kinematics equation and D-H variable representation. To drive the kinematics equation, we have to set up the standard coordinates first. The many links and the more complicated structure cause the difficult kinematics problem to solve in the geometrical way. Thus, we can solve the robot's kinematics problems efficiently and systematically by Denavit-Hartenberg's representation. Finally, with the coordinate values calculated above, it can draw a buffing gauge-line on the upper. Also, it can program off-line robot path on the shoes' upper. We are subjected to obtaining shoes' outline points, which are 2 outlines coupled with the points and the normal vector based on the points. These data is supposed to be transformed into .dxf file to be used for data of automatic buffing robot. This system developed is simulated by using spline curves coupled with each point from dxf file in Autocad. As a result of applying this system to the buffing robot in the flexible footwear manufacturing system, it can be used effectively to program the path of a real buffing robot.

  • PDF

Variable Rate IMBE-LP Coding Algorithm Using Band Information (주파수대역 정보를 이용한 가변률 IMBE-LP 음성부호화 알고리즘)

  • Park, Man-Ho;Bae, Geon-Seong
    • Journal of the Institute of Electronics Engineers of Korea SP
    • /
    • v.38 no.5
    • /
    • pp.576-582
    • /
    • 2001
  • The Multi-Band Excitation(MBE) speech coder uses a different approach for the representation of the excitation signal. It replaces the frame-based single voiced/unvoiced classification of a classical speech coder with a set of such decision over harmonic intervals in the frequency domain. This enables each speech segment to be a mixture of voiced and unvoiced, and improves the synthetic speech quality by reducing decision errors that might occur on the frame-based single voiced and unvoiced decision process when input speech is degraded with noise. The IMBE-LP, improved version of MBE with linear prediction, represents the spectral information of MBE model with linear prediction coefficients to obtain low bit rate of 2.4 kbps. In this Paper, we proposed a variable rate IMBE-LP vocoder that has lower bit rate than IMBE-LP without degrading the synthetic speech quality. To determine the LP order, it uses the spectral band information of the MBE model that has something to do with he input speech's characteristics. Experimental results are riven with our findings and discussions.

  • PDF

A Study on the types of PDA Icons and their Communication Capacity (개인용 정보단말기(PDA)에 사용되는 아이콘의 직관적 의미전달능력에 관한 연구)

  • 신명희
    • Archives of design research
    • /
    • v.17 no.2
    • /
    • pp.269-278
    • /
    • 2004
  • This research categorizes various icons that used in PDAs according to symbolization patterns, operating systems and support for color display. Since different icons vary in communication capacity I executed this research to verify it positively. In result, PDA loons were found to have different intuitive communication capacity according to its functions, symbolization pattern, operating system and use of color. \circled1 Icons which have similar object as ones that are used on desktop computer and icons with accurate, simple expression seems to have higher intuitive communication capacity among the icons categorized by functions. \circled2 Among the icons categorized by symbolization pattern, ones that express the action related to their functions have the highest recognition accuracy and longer delay before recognition. \circled3 Among the icons categorized by operating systems, ones that have concrete expression of object and a number of representation elements have higher recognition accuracy and longer delay before recognition. \circled4 Among the icons categorized by color and grayscale, ones with color have superior communication capacity due to additional stimulation although LCDs in most PDAs have limited color depth.

  • PDF